4673 Commits

Author SHA1 Message Date
kthelgason
29a44e351e This is a resubmission of https://codereview.webrtc.org/2047513002/
Original description:
Add proper lifetime of encoder-specific settings.

Permits passing VideoEncoderConfig between threads and not worry about
the lifetime of an underlying void pointer. Also adds type safety to
unpacking of codec-specific settings.

These settings are not yet propagating to VideoEncoder interfaces, but
the aim is to get rid of webrtc::VideoCodec for VideoEncoder.

BUG=webrtc:3424
R=perkj@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org

Review-Url: https://codereview.webrtc.org/2347843002
Cr-Commit-Position: refs/heads/master@{#14396}
2016-09-27 10:52:05 +00:00
minyue
c8299f9f87 Posting Opus's set-force-channels functionality to WebRTC.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2352713005
Cr-Commit-Position: refs/heads/master@{#14394}
2016-09-27 09:08:54 +00:00
danilchap
20e77c7b8a Unify rtcp packet setters
Renamed setters in rtcp classes
from WithField to SetField
from WithItem to AddItem or SetItems
from From to SetSenderSsrc
from To to SetMediaSsrc
Some redundant or unsued setters removed.
Pass-by-const& replaced with pass-by-value when appropriate.

BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/2348623003
Cr-Commit-Position: refs/heads/master@{#14393}
2016-09-27 08:37:51 +00:00
kjellander
4ecd9700ee GN: Fix incorrect include_dir for video_coding on iOS
When rtc_build_libyuv=false an incorrect code path
is surfaced in GN.

BUG=webrtc:6412
NOTRY=True
TESTED=gn gen out/foo --args='rtc_build_libyuv=false target_os="ios"'

Review-Url: https://codereview.webrtc.org/2375603002
Cr-Commit-Position: refs/heads/master@{#14392}
2016-09-27 08:11:24 +00:00
henrika
0a52c7003d THis CL enables possibility to select full-duplex OpenSL ES audio in AppRTCDemo, i.e., it adds support for OpenSL ES for input as well. The user must explicitly select this new mode in the debug UI hence it is not the default selection. There is no separate UI for input and output; instead both are enabled/disabled by the same switch.
BUG=webrtc:5925
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2366383002 .

Cr-Commit-Position: refs/heads/master@{#14390}
2016-09-27 07:35:37 +00:00
nisse
64ec8f826f Reland of Move MutableDataY{,U,V} methods to I420Buffer only. (patchset #1 id:1 of https://codereview.webrtc.org/2354223002/ )
Reason for revert:
Downstream application now fixed.

Original issue's description:
> Revert of Move MutableDataY{,U,V} methods to I420Buffer only. (patchset #14 id:260001 of https://codereview.webrtc.org/2278883002/ )
>
> Reason for revert:
> Broke downstream application.
>
> Original issue's description:
> > Move MutableDataY{,U,V} methods to I420Buffer only.
> >
> > Deleted from the VideoFrameBuffer base class.
> >
> > BUG=webrtc:5921
> >
> > Committed: https://crrev.com/5539ef6c03c273f39fadae41ace47fdc11ac6d60
> > Cr-Commit-Position: refs/heads/master@{#14317}
>
> TBR=perkj@webrtc.org,magjed@webrtc.org,pthatcher@webrtc.org,honghaiz@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5921
>
> Committed: https://crrev.com/776870a2599b8f43ad56987f9031690e3ccecde8
> Cr-Commit-Position: refs/heads/master@{#14325}

TBR=perkj@webrtc.org,magjed@webrtc.org,pthatcher@webrtc.org,honghaiz@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5921

Review-Url: https://codereview.webrtc.org/2372483002
Cr-Commit-Position: refs/heads/master@{#14389}
2016-09-27 07:17:40 +00:00
nisse
c637389949 Delete unused file mock_audio_vector.h.
BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2367323002
Cr-Commit-Position: refs/heads/master@{#14388}
2016-09-27 06:29:57 +00:00
stefan
89175a606e Trust that calls to RemoteEstimatorProxy::Process are done at the right frequency.
BUG=None

Review-Url: https://codereview.webrtc.org/2365293002
Cr-Commit-Position: refs/heads/master@{#14386}
2016-09-26 18:56:03 +00:00
minyue
fd8e33d3ad Removing a useless ctor in AudioNetworkAdaptorImpl.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2367743002
Cr-Commit-Position: refs/heads/master@{#14384}
2016-09-26 18:46:45 +00:00
kjellander
464382da71 Remove duplicated entry for bwe_simulations.cc
Since modules_unittests already depends on
remote_bitrate_estimator:bwe_simulator and the bwe_simulations.cc
source was added to that target in https://codereview.webrtc.org/2296253002
there's no point having it added here.

BUG=webrtc:6323
NOTRY=True
NOPRESUBMIT=True
NOTREECHECKS=True

Review-Url: https://codereview.webrtc.org/2368933002
Cr-Commit-Position: refs/heads/master@{#14380}
2016-09-26 10:00:09 +00:00
zijiehe
c59bf0415a Remove differ from ScreenCapturer implementations
We can use ScreenCapturerDifferWrapper if needed, otherwise ScreenCapturer does
not need to calculate updated region itself, setting to entire screen is enough.

BUG=633802

Review-Url: https://codereview.webrtc.org/2348803003
Cr-Commit-Position: refs/heads/master@{#14377}
2016-09-24 00:54:40 +00:00
solenberg
d3d230f788 - Make RtpSenderAudio not inherit from DtmfQueue.
- Remove unused method DtmfQueue::ResetDTMF()

BUG=webrtc:2795

Review-Url: https://codereview.webrtc.org/2365873002
Cr-Commit-Position: refs/heads/master@{#14376}
2016-09-23 20:10:50 +00:00
danilchap
92ea601e90 Move class RTCPHelp::RTCPPacketInformation into RTCPReceiver
Use it by pointer instead of by reference.
Renamed PacketInformation members to follow style,
Unused members removed.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2366563002
Cr-Commit-Position: refs/heads/master@{#14375}
2016-09-23 17:36:12 +00:00
henrika
dda366611e Fixes minor issue in AudioDeviceTest.RunPlayoutAndRecordingInFullDuplex for iOS.
Followup on https://codereview.webrtc.org/2349263004/

BUG=NONE
NOTRY=TRUE

Review-Url: https://codereview.webrtc.org/2362263002
Cr-Commit-Position: refs/heads/master@{#14374}
2016-09-23 15:42:49 +00:00
magjed
44428a8aa6 iOS: Always build H264 HW encoder/decoder
This CL removes the use_objc_h264 flag. This means that the VideoToolbox
H264 encoder and decoder will always be built.

BUG=webrtc:4081
NOTRY=TRUE

Review-Url: https://codereview.webrtc.org/2366443003
Cr-Commit-Position: refs/heads/master@{#14372}
2016-09-23 14:01:44 +00:00
ossu
f1b08da5b4 Stopped using the NetEqDecoder enum internally in NetEq.
NetEqDecoder is still used in the external interfaces, but this change
opens up the ability to use SdpAudioFormats directly, once appropriate
interfaces have been added.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2355503002
Cr-Commit-Position: refs/heads/master@{#14368}
2016-09-23 09:19:49 +00:00
asapersson
1490f7aa55 Add histogram for end-to-end delay:
"WebRTC.Video.EndToEndDelayInMs"

Make capture time in local timebase available for decoded VP9 video frames (propagate ntp_time_ms from EncodedImage to decoded VideoFrame).

BUG=webrtc:6409

Review-Url: https://codereview.webrtc.org/1905563002
Cr-Commit-Position: refs/heads/master@{#14367}
2016-09-23 09:09:59 +00:00
minyue
6d4c8c307e Renaming a proto target in GYP for audio network adaptor.
It was incorrectly named for GYP in https://codereview.webrtc.org/2365723002
This makes the target name be the same for GN and GYP.

BUG=webrtc:6303
NOTRY=True

Review-Url: https://codereview.webrtc.org/2366883002
Cr-Commit-Position: refs/heads/master@{#14366}
2016-09-23 08:42:22 +00:00
kjellander
b62dbbe985 GN: Change rtc_source_set targets --> rtc_static_library
This changes most non-test related rtc_source_set targets to be
rtc_static_library instead. Targets without any .cc files are excluded.
This should bring back the build behavior we used to have with GYP
(i.e. same symbols exported in the libjingle_peerconnection.a file, which
are used by some downstream projects).

After doing an Android build with these changes:
$ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf
00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory
$ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf
00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*)
00000001 T webrtc::CreatePeerConnectionFactory()

See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries
for more details on this.

NOTICE: This should be further cleaned up in the future, to reduce
binary bloat and unnecessary linking time. Right now it's more
important to restore the desired build output though.

BUG=webrtc:6410, chromium:630755

Review-Url: https://codereview.webrtc.org/2361623004
Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 07:38:58 +00:00
minyue
25f6a39181 Relanding of "Adding debug dump to audio network adaptor."
The original CL was https://codereview.webrtc.org/2356763002

but got reverted https://codereview.webrtc.org/2362003002/.

The error was that ana_debug_dump_proto as a proto_library was placed under rtc_include_tests.

BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2365723002
Cr-Commit-Position: refs/heads/master@{#14363}
2016-09-23 05:23:28 +00:00
minyue
161b3907ab Revert of Adding debug dump to audio network adaptor. (patchset #5 id:140001 of https://codereview.webrtc.org/2356763002/ )
Reason for revert:
Chromium bot fails

Original issue's description:
> Adding debug dump to audio network adaptor.
>
> BUG=webrtc:6303
>
> Committed: https://crrev.com/7e4f8928062afc8d571bb69f3223711701cbaad6
> Cr-Commit-Position: refs/heads/master@{#14361}

TBR=michaelt@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2362003002
Cr-Commit-Position: refs/heads/master@{#14362}
2016-09-22 21:17:01 +00:00
minyue
7e4f892806 Adding debug dump to audio network adaptor.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2356763002
Cr-Commit-Position: refs/heads/master@{#14361}
2016-09-22 20:39:18 +00:00
henrika
051d151569 Adds audio session status to logs for each valid audio route change on iOS
BUG=b/30944297
NOTRY=TRUE

Review-Url: https://codereview.webrtc.org/2355393005
Cr-Commit-Position: refs/heads/master@{#14355}
2016-09-22 15:48:10 +00:00
kthelgason
c37e9835a7 Add custom info.plist to modules_unittests
This is to fix an issue introduced with iOS 10 where all applications that access the microphone have to include a string in the Info.plist file explaining why they need it.

BUG=webrtc:6403

Review-Url: https://codereview.webrtc.org/2359863003
Cr-Commit-Position: refs/heads/master@{#14354}
2016-09-22 15:00:57 +00:00
danilchap
f292e31511 Relax too strict DCHECKs while parsing rtcp reports
BUG=chromium:649129

Review-Url: https://codereview.webrtc.org/2361493004
Cr-Commit-Position: refs/heads/master@{#14353}
2016-09-22 14:24:38 +00:00
minyue
d0ede4493e Adding FecController to audio network adaptor.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2337103006
Cr-Commit-Position: refs/heads/master@{#14351}
2016-09-22 13:20:59 +00:00
danilchap
799a9d017a Revert of Remove unnecessary interface TelephoneEventHandler (patchset #3 id:40001 of https://codereview.webrtc.org/2357583002/ )
Reason for revert:
breaks downstream code

Original issue's description:
> Remove unnecessary interface TelephoneEventHandler.
>
> BUG=webrtc:2795
>
> Committed: https://crrev.com/2beb42983ca24e1326a9a7f2c06b3ad740eea2c3
> Cr-Commit-Position: refs/heads/master@{#14346}

TBR=henrik.lundin@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:2795

Review-Url: https://codereview.webrtc.org/2362673002
Cr-Commit-Position: refs/heads/master@{#14348}
2016-09-22 10:36:34 +00:00
ossu
a70695a3e1 Moved Opus-specific payload splitting into AudioDecoderOpus.
The biggest change to NetEq is the move from a primary flag, to a
Priority with two separate levels: one set by RED splitting and one
set by the codec itself. This allows us to unambigously prioritize
"fallback" packets from these two sources. I've chosen what I believe
is the sensible ordering: packets that the codec prioritizes are
chosen first, regardless of if they are secondary RED packets or
not. So if we were to use Opus w/ FEC in RED, we'd only do Opus FEC
decoding if there was no RED packet that could cover the time slot.

With this change, PayloadSplitter now only deals with RED
packets. Maybe it should be renamed RedPayloadSplitter?

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2342443005
Cr-Commit-Position: refs/heads/master@{#14347}
2016-09-22 09:07:03 +00:00
solenberg
2beb42983c Remove unnecessary interface TelephoneEventHandler.
BUG=webrtc:2795

Review-Url: https://codereview.webrtc.org/2357583002
Cr-Commit-Position: refs/heads/master@{#14346}
2016-09-22 08:46:08 +00:00
minyue
bc77ed7657 Adding reordering logic in audio network adaptor.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2349113002
Cr-Commit-Position: refs/heads/master@{#14344}
2016-09-22 07:45:23 +00:00
minyue
4aec1d4437 Relanding of "Adding BitrateController to audio network adaptor."
Adding BitrateController to audio network adaptor was first landed in https://codereview.webrtc.org/2334613002/ but reverted in https://codereview.webrtc.org/2352223002/ due to ODR violation.

This CL tries to use namespace trick to solve the ODR problem.

BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2353293002
Cr-Commit-Position: refs/heads/master@{#14343}
2016-09-22 06:01:34 +00:00
zijiehe
05bba2be48 Several minor changes to ScreenCapturerWinMagnifier
1. Remove legacy screen-saver-blocking logic
2. tls_index_ is not a good choice, we can use thread-static
3. ScreenCapturerHelper is not designed for this scenario
4. Disable this capturer on 2+ monitors system

BUG=638802

Review-Url: https://codereview.webrtc.org/2319383002
Cr-Commit-Position: refs/heads/master@{#14342}
2016-09-22 00:25:48 +00:00
zijiehe
66cadfc0e3 Several minor improvements of DirectX capturer
1. It looks like ComPtr cannot work well with vector::emplace_back, I got a
consistent crash on one of my machine, but not the other. Move constructor
should have no impact to lvalue reference, but I may be wrong here. The
impact here is ComPtr released before it should be. So a simple solution is to
use copy instead of reference. The D3dDevice is a collection of reference
counted pointers (Microsoft::WRL::ComPtr), there is almost no extra cost.

2. Actively set several fields in D3D11_TEXTURE2D_DESC to avoid potential break
if there are some platform changes later.

3. AcquireNextFrame returns both a DXGI_OUTDUPL_FRAME_INFO with
AccumulatedFrames and an IDXGIResource. But there is no comment in MSDN to
ensure IDXGIResource won't be nullptr if AccumulatedFrames > 0. Adding an extra
check in DxgiOutputDuplicator makes it a safer.

BUG=314516

Review-Url: https://codereview.webrtc.org/2345163002
Cr-Commit-Position: refs/heads/master@{#14341}
2016-09-22 00:19:15 +00:00
qiangchen
6f79d840ba Bug Fix: Mac Retina Screen Capture's Mouse Cursor Too Small
On retina display, when we do screen capture, the mouse cursor
looks too small. The reason is that we painted the cursor with
low resolution on to the frame directly.

This CL fixes the bug.

BUG=632995

Review-Url: https://codereview.webrtc.org/2350743003
Cr-Commit-Position: refs/heads/master@{#14340}
2016-09-21 23:45:03 +00:00
minyue
e35d329315 Adding FrameLengthController to audio network adaptor.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2335163002
Cr-Commit-Position: refs/heads/master@{#14339}
2016-09-21 23:00:38 +00:00
zijiehe
acc39c44bc Use RgbaColor in DesktopFrameGenerator and add RgbaColorTest
This change uses RgbaColor in DesktopFrameGenerator instead of raw uint32_t to
avoid potential endian issues.

BUG=633802

Review-Url: https://codereview.webrtc.org/2334853002
Cr-Commit-Position: refs/heads/master@{#14337}
2016-09-21 19:23:22 +00:00
kwiberg
c4ccd4d61c AcmReceiver: Eliminate AcmReceiver::decoders_
BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/2351183002
Cr-Commit-Position: refs/heads/master@{#14335}
2016-09-21 17:55:21 +00:00
henrika
c5aea65b76 Adds output audio volume to iOS logs
BUG=b/30944297
NOTRY=TRUE

Review-Url: https://codereview.webrtc.org/2360583002
Cr-Commit-Position: refs/heads/master@{#14334}
2016-09-21 14:46:01 +00:00
ossu
7f40ba4414 Moved legacy_encoded_audio_frame into audio_decoder_interface.
audio_decoder.cc depends on LegacyEncodedAudioFrame and
LegacyEncodedAudioFrame depends on AudioDecoder::EncodedAudioFrame, so
there's no clear way to separate them as of now. This error is also
hodling up builds downstream. I expect we'll revisit these
dependencies as part of the upcoming larger restructuring effort.

NOTRY=true
BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2359763002
Cr-Commit-Position: refs/heads/master@{#14329}
2016-09-21 12:50:45 +00:00
henrika
17802ae258 Ensures that ADM for Android and iOS uses identical states when stopping audio
BUG=b/25975010
TBR=tkchin
NOTRY=TRUE

Review-Url: https://codereview.webrtc.org/2349263004
Cr-Commit-Position: refs/heads/master@{#14328}
2016-09-21 11:55:10 +00:00
minyue
33b96b3588 Revert of Adding BitrateController to audio network adaptor. (patchset #7 id:140001 of https://codereview.webrtc.org/2334613002/ )
Reason for revert:
ODR violation

Original issue's description:
> Adding BitrateController to audio network adaptor.
>
> BUG=webrtc:6303
>
> Committed: https://crrev.com/26b039a137be0a8703766f45b546b29323de714f
> Cr-Commit-Position: refs/heads/master@{#14293}

TBR=michaelt@webrtc.org,henrik.lundin@webrtc.org,krasin@chromium.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2352223002
Cr-Commit-Position: refs/heads/master@{#14327}
2016-09-21 11:30:23 +00:00
nisse
776870a259 Revert of Move MutableDataY{,U,V} methods to I420Buffer only. (patchset #14 id:260001 of https://codereview.webrtc.org/2278883002/ )
Reason for revert:
Broke downstream application.

Original issue's description:
> Move MutableDataY{,U,V} methods to I420Buffer only.
>
> Deleted from the VideoFrameBuffer base class.
>
> BUG=webrtc:5921
>
> Committed: https://crrev.com/5539ef6c03c273f39fadae41ace47fdc11ac6d60
> Cr-Commit-Position: refs/heads/master@{#14317}

TBR=perkj@webrtc.org,magjed@webrtc.org,pthatcher@webrtc.org,honghaiz@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5921

Review-Url: https://codereview.webrtc.org/2354223002
Cr-Commit-Position: refs/heads/master@{#14325}
2016-09-21 10:52:21 +00:00
Rasmus Brandt
ea7beb9741 Reorder member functions in RtpFecTest.
Place member functions before tests. No changes to the functionality.

BUG=webrtc:5654
R=danilchap@webrtc.org

Review URL: https://codereview.webrtc.org/2297533002 .

Cr-Commit-Position: refs/heads/master@{#14322}
2016-09-21 10:01:30 +00:00
philipel
1f39ba1cd9 Copy payload data when inserting packets into video_coding::PacketBuffer.
The payload pointed to by |dataPtr| is volatile and needs to be copied
to its own buffer.

BUG=webrtc:5514
R=brandtr@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2302763002 .

Cr-Commit-Position: refs/heads/master@{#14321}
2016-09-21 09:27:56 +00:00
ossu
0d526d558b Moved codec-specific audio packet splitting into decoders.
There's still some code run specifically for Opus w/ FEC. It will be
addressed in a separate CL.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2326003002
Cr-Commit-Position: refs/heads/master@{#14319}
2016-09-21 08:57:36 +00:00
nisse
5539ef6c03 Move MutableDataY{,U,V} methods to I420Buffer only.
Deleted from the VideoFrameBuffer base class.

BUG=webrtc:5921

Review-Url: https://codereview.webrtc.org/2278883002
Cr-Commit-Position: refs/heads/master@{#14317}
2016-09-21 08:27:38 +00:00
Rasmus Brandt
78db1582e5 Generalize FEC header formatting.
- Split out reading/writing of FEC headers to classes separate
  from ForwardErrorCorrection. This makes ForwardErrorCorrection
  oblivious to what FEC header scheme is used, and lets it focus on
  encoding/decoding the FEC payloads.
- Add unit tests for FEC header readers/writers.
- Split ForwardErrorCorrection::XorPackets into XorHeaders and
  XorPayloads and reuse these functions for both encoding and
  decoding.
- Rename AttemptRecover -> AttemptRecovery in ForwardErrorCorrection.

BUG=webrtc:5654
R=danilchap@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2260803002 .

Cr-Commit-Position: refs/heads/master@{#14316}
2016-09-21 07:19:42 +00:00
brandtr
ece4aba64e Generalize FEC unit tests and rename GenerateFec.
- Rename GenerateFec -> EncodeFec in ForwardErrorCorrection. This naming
  is more consistent with DecodeFec.
- Add appropriate using directives, to reduce clutter in tests.
- Move ConstructMediaPackets to fec_test_helper.{h,cc}. This will help
  future tests of ULPFEC/FlexFEC header formatters.
- Generalize tests in rtp_fec_unittest.cc to typed tests. This will help
  testing ForwardErrorCorrection with both ULPFEC and FlexFEC.

This CL should not impact functionality or performance.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2267393002
Cr-Commit-Position: refs/heads/master@{#14314}
2016-09-21 06:16:36 +00:00
minyue
3548357e1b Adding SmoothingFilter to audio network adaptor.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2339523002
Cr-Commit-Position: refs/heads/master@{#14313}
2016-09-21 06:13:16 +00:00
kwiberg
d120192f32 AcmReceiver::DecoderByPayloadType: Ask NetEq for decoder
Instead of looking in AcmReceiver::decoders_, which we're trying to
get rid of.

(This is a re-land of https://codereview.webrtc.org/2341283002.)

BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/2352623002
Cr-Commit-Position: refs/heads/master@{#14312}
2016-09-20 22:18:24 +00:00